Manual Grandstream

5/21/2018 Manual Grandstream

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Grandstream Networks, Inc. GXP User Manual Page1 of 44

Firmware 1.2.5.3 Last Updated: 03/2011

Grandstream Networks, Inc.GXP2020/GXP2010/GXP2000/GXP1200/GXP280/GXP285Enterprise IP Phones

GXP-2020 GXP-2010

GXP-1200 GXP-280/GXP-285GXP-2000


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Grandstream Networks, Inc. GXP User Manual Page2 of 44Firmware 1.2.5.3 Last Updated: 03/2011

TABLE OF CONTENTSGXP USER MANUAL

WELCOME ................................................................................................................................................................. 4

INSTALLATION......................................................................................................................................................... 5

EQUIPMENT PACKAGING ............................................................................................................................................. 5

CONNECTING YOUR PHONE ........................................................................................................................................ 5

GXP-2000EXTENSION UNIT....................................................................................................................................... 5

SAFETY COMPLIANCES ................................................................................................................................................ 7

WARRANTY ................................................................................................................................................................. 7

PRODUCT OVERVIEW ............................................................................................................................................ 8

USING THE GXP SIP ENTERPRISE PHONE ..................................................................................................... 13

GETTING FAMILIAR WITH THE LCD .......................................................................................................................... 13

MAKING PHONE CALLS ............................................................................................................................................. 17

ANSWERING PHONE CALLS ....................................................................................................................................... 20

PHONE FUNCTIONS DURING A PHONE CALL ............................................................................................................. 20

CALL FEATURES ........................................................................................................................................................ 23

CUSTOMIZED LCDSCREEN &XML ......................................................................................................................... 23

CONFIGURATION GUIDE ................................................................................................ ...................................... 24

CONFIGURATION VIA KEYPAD .................................................................................................................................. 24

CONFIGURATION VIAWEB BROWSER ...................................................................................................................... 27

SAVING THE CONFIGURATION CHANGES ................................................................................................................... 41

REBOOTING THE PHONE REMOTELY ......................................................................................................................... 41

SOFTWARE UPGRADE & CUSTOMIZATION .................................................................................................. 42

FIRMWARE UPGRADE THROUGH TFTP/HTTP .......................................................................................................... 42

CONFIGURATION FILE DOWNLOAD ........................................................................................................................... 43

RESTORE FACTORY DEFAULT SETTING ....................................................................................................... 44

TABLE OF FIGURESGXP USER MANUAL

Figure 1: Connecting the GXP2000 and the GXPExtension ...................................................... 6Figure 2: GXP2000 Internal Headset Wiring Schema .................................................................. 7Table 10: GXP Keypad Buttons .................................................................................................... 16Figure 3: Keypad GUI Flow ............................................................................................................ 26

TABLE OF TABLESGXP USER MANUAL

Table 1: Equipment Packaging ....................................................................................................... 5Table 2: GXP Connectors ............................................................................................................... 5Table 3: GXP Product Models ......................................................................................................... 8Table 4: GXP Comparison Guide .................................................................................................... 9Table 5: GXP Key Features in a Glance ......................................................................................... 9Table 6: GXP Hardware Specifications

......................................................................................... 10


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Table 7: GXP Technical Specifications ......................................................................................... 11Table 8: LCD Buttons

.................................................................................................................... 13Table 9: LCD Icons

....................................................................................................................... 14Table 11: GXP Call Features ........................................................................................................ 23Table 12: Key Pad Configuration Menu

........................................................................................ 24Table 13: Device Configuration - Status

....................................................................................... 28Table 14: Device Configuration Basic Settings

.......................................................................... 28Table 15: Advanced Settings ........................................................................................................ 31Table 16: SIP Account Settings .................................................................................................... 37

GUI INTERFACE EXAMPLESGXP USER MANUAL

(http://www.grandstream.com/support/gxp_series/general/documents/gxp_gui.zip)

1. SCREENSHOT OF CONFIGURATION LOGIN PAGE

2. SCREENSHOT OF STATUS PAGE

3. SCREENSHOT OF BASIC SETTINGS CONFIGURATION PAGE

4. SCREENSHOT OFADVANCED USER CONFIGURATION PAGE

5. SCREENSHOT OF SIPACCOUNT CONFIGURATION

6. SCREENSHOT OF SAVED CONFIGURATION CHANGES

7. SCREENSHOT OF REBOOT PAGE
http://www.grandstream.com/support/gxp_series/general/documents/gxp_gui.ziphttp://www.grandstream.com/support/gxp_series/general/documents/gxp_gui.ziphttp://www.grandstream.com/support/gxp_series/general/documents/gxp_gui.ziphttp://www.grandstream.com/support/gxp_series/general/documents/gxp_gui.zip


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Welcome

Your Grandstream GXP Series IP phone features a new sophisticated design and is very easy to use.The GXP combines advanced feature functionality with the latest technology to offer excellent audioquality, ease of use, expandability, and broad interoperability with 3rd party SIP platforms. It is ideal for

the enterprise customer.

The GXP Series supports a broad range of codecs, security protection, PoE (not supported on GXP-280),dual 10/100mbps Ethernet ports and are very easy to manage. Currently, the GXP Series consists of thefollowing six models: GXP-280, GXP-285, GXP-1200, GXP-2000, GXP-2010 and GXP-2020. Each modeldelivers superior audio quality using either a handset, hands-free speakerphone or headset (except forGXP2000) and supports multi-party conferencing, multi-languages, dual-color LEDs, presence and BLF(on most models). Large easy-to-read backlit graphical displays with multiple XML keys further enhancethe user experience (not supported on GXP-280/285). Some models (GXP-2000, GXP2010 andGXP2020 currently) are expandable with one or two expansion module.

The series is based on SIP standard and are interoperable with most 3rd party SIP platforms and open-source platforms.

Caution:Changes or modifications to this product not expressly approved by Grandstream, or operationof this product in any way other than as detailed by this User Manual, could void your manufacturerwarranty.

Warning: Please do not use a different power adaptor with the GXP as it may cause damage to theproducts and void the manufacturer warranty.

This document is contains links to Grandstream GUI Interfaces. Please download these examples

fromhttp://www.grandstream.com/support/gxp_series/general/documents/gxp_gui.zipfor yourreference.

This document is subject to change without notice. The latest electronic version of this user manualis available for download@:http://www.grandstream.com/support/gxp_series/general/documents/gxp_usermanual_english.pdf

Reproduction or transmittal of the entire or any part, in any form or by any means, electronic or print,for any purpose without the express written permission of Grandstream Networks, Inc. is notpermitted.
http://www.grandstream.com/support/gxp_series/general/documents/gxp_gui.ziphttp://www.grandstream.com/support/gxp_series/general/documents/gxp_gui.ziphttp://www.grandstream.com/support/gxp_series/general/documents/gxp_gui.ziphttp://www.grandstream.com/support/gxp_series/general/documents/gxp_usermanual_english.pdfhttp://www.grandstream.com/support/gxp_series/general/documents/gxp_usermanual_english.pdfhttp://www.grandstream.com/support/gxp_series/general/documents/gxp_usermanual_english.pdfhttp://www.grandstream.com/support/gxp_series/general/documents/gxp_usermanual_english.pdfhttp://www.grandstream.com/support/gxp_series/general/documents/gxp_gui.zip


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Installation

EQUIPMENT PACKAGING

Table 1: Equipment Packaging

GXP-280/285 GXP-1200 GXP-2000 GXP-2010 GXP-2020Main Case Yes Yes Yes Yes YesHandset Yes Yes Yes Yes YesPhone Cord Yes Yes Yes Yes YesPower Adaptor Yes Yes Yes Yes YesEthernet Cable Yes Yes Yes Yes YesHigh Phone Stand No Yes No Yes YesLow Phone Stand Yes No No Yes YesWall Mount Spacers (2) No Yes No Yes Yes

CONNECTING YOUR PHONE

The connectors of the GXP1200/2010/2020 are located on the bottom of the device while they are located onthe back side of the GXP280/285/2000.

Table 2: GXP Connectors

EXTConnects the GXP Extension unit directly to the GXP using connection cable.Draws power from PoE if provided by network.

PC 10/100Mbps RJ-45 ports for PC (downlink) connection.

LAN10/100Mbps RJ-45 port for LAN (uplink) connection. Supports PoE (802.3af).Draws power from either spare line or signal line.

Power Jack 5V DC power port; UL Certified

Headset JackRJ22 and 2.5mm for GXP-280/285/2010/2020RJ22 for GXP-12002.5mm for GXP-2000 HW Rev1.0 or later

Handset Jack RJ11

GXP-2000EXTENSION UNIT

GXP2000 supports two (2) extension units, providing up to 112 additional programmable extensions. EachGXP Extension unit has 56 multipurpose keys, dual color LEDs (red/green) and support BLF (Busy LampField) and Presence.

GXP2000 Extension package contains:


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1) One GXP Extension unit

2) One PS2 cable3) One connection plate4) One Universal Power Adaptor

FIGURE 1: CONNECTING THE GXP2000AND THE GXPEXTENSION

GXP2000 w/GXPExtension GXP Extension

Connecting the GXP2000w/GXPExtension

Reverse side of connectionw/connection p late

Connect the first GXP EXT to the GXP2000 using the PS2 cable found in the GXP Extension package.The first GXPExt draws power directly from the phone. Connect the second GXP Extension unit using theconnection plate and the PS2 cable. The GXP2000 will automatically reboot and power up the GXPExtensions. Grandstream recommends, though not required, to use a separate power supply with thesecond GXP Ext.

NOTE: should your system lose power, please unplug your devices and power up the GXP2000 first.

Powering up the system:

1. The GXP2000 will boot up first;2. The GXP LEDs will be solid red;3. The status light in the top right corner of the GXPExt will blink red;4. All of the LED indicators on the GXPExt will flash three times;5. The status light at the top right corner of the GXPExt will turn to solid green.

NOTE:1. Extension for GXP2010 and GXP2020 does not support hot-swap. Once connected, user shouldreboot the phone to ensure the set up will work correctly.

2. GXP2010/GXP2020 can drive 2 extension modules. Independent power adapters are not neededfor extension modules.


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Figure 2: GXP2000 Internal Headset Wiring Schema

NOTE:For GXP-2000 HW REV. 0.3 and 0.4, a 3.5mm to 2.5mm plug converter is required to use a 2.5mmheadset. The converter can be purchased at any electronics store.

SAFETY COMPLIANCES

The GXP phone complies with FCC/CE and various safety standards. The GXP power adaptor is compliantwith the UL standard. Only use the universal power adaptor provided with the GXP package. Themanufacturers warranty does not cover damages to the phone caused by unsupported power adaptors.

WARRANTY

If you purchased your GXP from a reseller, please contact the company where you purchased your phonefor replacement, repair or refund. If you purchased the product directly from Grandstream, contact yourGrandstream Sales and Service Representative for a RMA (Return Materials Authorization) number beforeyou return the product. Grandstream reserves the right to remedy warranty policy without prior notification.


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Product Overview

Table 3: GXP Product Models

Model Picture Overview

GXP-280

GXP280 is an entry-level SIP phone. It features:

Single line Three soft keys

GXP-285


Single line Three soft keys PoE Supported

GXP-1200


Two lines Three soft keys PoE Supported

GXP-2000

GXP2000 is a mainstream SIP phone. It features:

Four lines Seven programmable hard keys PoE Supported

GXP-2010

GXP2010 is a key system SIP phone. It features:

Four lines Eighteen programmable hard keys Three XML programmable soft keys PoE Supported


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GXP-2020

GXP2020 is an executive SIP phone. It features:

Six lines Seven programmable hard keys Four XML programmable soft keys PoE Supported

Table 4: GXP Comparison Guide

Features GXP-280/285 GXP-1200 GXP-2000 GXP-2010 GXP-2020

LCD Display 128x32 pixel 128x32 pixel 130x64 pixel 240x120 pixel 320x160 pixel

Number of Lines 1 2 4 4 6

ProgrammableHard Keys

No No 7 18 7

Soft Keys 3 3 No 3 4

Extension Module No No Yes, up to 2ExpansionModules, 56nodes each

Yes, up to 2ExpansionModules, 56nodes each

Yes, up to 2ExpansionModules, 56nodes each

Table 5: GXP Key Features in a Glance

Features Benefits

Open Standards Compatible SIP 2.0, TCP/IP/UDP, RTP/RTCP, HTTP/HTTPS, ARP/RARP, ICMP,DNS (A record and SRV), DHCP (both client and server), PPPoE,TFTP, NTP, Telnet.

Superb Audio Quality Advanced Digital Signal Processing (DSP), Silence suppression,VAD, CNG, AGC.

Network Interfaces Dual 10/100mbps Ethernet ports, headset jack (RJ22 and/or 2.5mmjack).

Feature Rich Traditional voice features including caller ID, call waiting, hold,transfer, forward, block, and off-hook dial, click to dial

Advanced Features Multi-line support with dual-color LED (except on GXP-280/285),multi-party conferencing, line extension interface, large back-lit(except on GXP-280/285) graphic LCD, 5 or 3 navigation keys,

dedicated buttons for hold, send, speakerphone, headset, transfer,conference (for up to 5 parties depending on model), mute, message,Do-not-disturb, phone book, intercom/paging.

Advanced Funct ional ity Custom downloadable ring-tones, SRTP, multi-language support andXML enabled, adjustable positioning angles, wall mountable, AESencryption.


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Table 6: GXP Hardware Specifications

LAN Interface (Ethernetports)

Two (2) 10/100 Mbps Full/Half Duplex Ethernet Switch with LAN and PC port withauto detection

Graphic LCD Display GXP-280/285 GXP-1200 GXP-2000 GXP-2010 GXP-2020

128x32

pixel

128x32 pixel 130x64pixel

240x120pixel

320x160pixel

Expansion Module SupportGXP-280/285 GXP-1200 GXP-2000 GXP-2010 GXP-2020

No No Yes Yes Yes

Headset JackGXP-280/285 GXP-1200 GXP-2000 GXP-2010 GXP-2020

2.5mm andRJ22

RJ22 2.5mm 2.5mm andRJ22

2.5mm andRJ22

Call Appearance LED Dual color (green/red)

GXP-280/285 GXP-1200

GXP-2000 GXP-2010 GXP-2020

No 3 11 22 13

Power over Ethernet Built-in auto-sensing: Cisco and IEEE 802.3af standard: phone drawspower from Ethernet (except on GXP-280)

Universal Switching Input: 100-240VAC 50-60 Hz

Power Adaptor Output: +5VDC, 1200mA,ULcertified

Dimension

GXP-280/285 168mm(l) x 200mm(w) x 89.5mm(h)

GXP-1200 210mm(l) x 195mm(w) x 77mm(h)

GXP-2000 220mm(l) x 215mm(w) x 57mm(h)

GXP-2010 210mm(l) x 250mm(w) x 77mm (h)

GXP-2020 251mm(l) x 202mm(w) x 77mm(h)

Weight

GXP-280/285 GXP-1200 GXP-2000 GXP-2010 GXP-2020

0.62kg

(1.37lbs)

0.86kg(1.91lbs)

0.82kg(1.81lbs)

1.1kg(2.44lbs)

1.66kg(3.64lbs)

Temperature 32 104F/ 0 40

C

Humidity 10% 90% (non-condensing)Compliance FCC / CE / C-Tick


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Table 7: GXP Technical Specifications

Lines Multiple direct lines with independent SIP accounts, programmable speed dial keys,XML programmable soft-keys (non programmable on GXP-280/285, GXP1200, GXP2000).

ProtocolSupport

Support SIP 2.0, TCP/UDP/IP, PPPoE, RTP/RTCP, SRTP by SDES, HTTP, ARP/RARP,ICMP, DNS, DHCP, NTP, TFTP, SIMPLE/PRESENCE protocolsSupports multiple SIP accounts

Supports SIP PUBLISH method (RFC 3903), SIP Presence package (RFC 3856, 3863) foruse of 7 MFKs, SIP Dialog package (RFC 4235)

Supports SIP MESSAGE method (RFC 3428)Stores up to 100 incoming IM messages

Display Back-lit graphic LCD display. (GXP-280/285 display is not back-lit)

FeatureKeys

GXP-280/285 GXP-1200 GXP-2000 GXP-2010 GXP-2020

HOLD Yes Yes Yes Yes Yes

SPEAKERPHONE Yes Yes Yes Yes Yes

SEND Yes Yes Yes Yes YesTRANSFER Yes Yes Yes Yes Yes

CONF Yes Yes Yes Yes Yes

MUTE Yes Yes Yes Yes Yes

DND Yes Yes Yes Yes Yes

HEADSET Yes Yes Yes Yes Yes

INTERCOM No No No Yes Yes

PHONEBOOK No No No Yes Yes

MSG Yes Yes Yes Yes Yes

MENU Yes Yes Yes Yes Yes

NAVIGATION (4) Yes (3) Yes Yes Yes Yes

Device

ManagementNAT-friendly remote software upgrade (via TFTP/HTTP) for deployed devices includingbehind firewall/NAT

Auto/manual provisioning system, GUI Interface

Support Layer 2 (802.1Q, VLAN, 802.1p) and Layer 3 QoS (ToS, DiffServ, MPLS)

Expansion interface, Address Book

AudioFeatures

Full-duplex hands-free speakerphone, headset enabledAdvanced Digital Signal Processing (DSP)

Dynamic negotiation of codec and voice payload length

Support for G.723,1 (5.3/6.3K), G.729A/B, G.711 a/-law, G.726-32, G.722 (wide-band),GSM and iLBC codecs

In-band and out-of-band DTMF (in audio, RFC2833, SIP INFO)

Silence Suppression, VAD (voice activity detection), CNG (comfort noise generation), AGC(automatic gain control)

Acoustic Echo Cancellation (AEC) with Automatic Gain Control (AGC) for speakerphonemode, Support side tone

Adaptive jitter buffer control and packet delay & loss concealment

Telephony

Features

Intuitive graphic user interface (GUI), downloadable phone book (XML, LDAP), support foranonymous call using privacy header, MLS (multi language support)

Voice mail indicator, downloadable custom ring-tones, call hold, call transfer


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(attended/blind), call forward, call waiting, caller ID, mute, redial, call log, caller ID display orblock, Do-Not-Disturb (DND) and volume control

Multi-party conferencing (up to 5), dial plan prefix, off-hook auto dial, auto answer, early dialand speed dial (on some models)

Network and

Provisioning

Via keypad/LCD, Web browser, or secure (AES encrypted) central configuration file, manual

or dynamic host configuration protocol (DHCP) network setupSupport NAT traversal using IETF STUN and Symmetric RTP

Support for IEEE 802.1p/Q tagging (VLAN), Layer 3 TOS

Firmware

Upgrades

Support firmware upgrade via TFTP or HTTP,

Support for Authenticating configuration file before accepting changes

User specific URL for configuration file and firmware files

AdvancedServerFeatures

Message waiting indication, support DNS SRV Look up and SIP Server Fail Over, Supportcustomizable idle screen via downloading XML by HTTP/TFTP

Security DIGEST authentication and encryption using MD5 and MD5-sess, SRTP


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Using the GXP SIP Enterprise Phone

GETTING FAMILIAR WITH THE LCD

GXP-2xxx has a dynamic and customizable screen. The screen displays differently depending on whetherthe phone is idle or in use (active screen).

Table 8: LCD Buttons

Key Button Key Button Definitions

LINESELECTORS

Selects the phone line printed on its right-hand side.

SIP PHONELINES

Displays the available phone lines. Choose a phone line by pressing the corresponding lineselector on the left-hand side.

DATE ANDTIME

Displays the current date and time. Can be synchronized with Internet time servers.

LOGODisplays company logo. This logo can be customized. For more information on customizing thelogo, please check page 24.

NETWORKSTATUS

Shows the status of the phone and network. It will indicate whether the network is down, starting oris running (show IP address). Other messages such as DO NOT DISTURB or ## MISSEDCALLS are shown here too.

STATUSBAR

Shows the status of the phone, using icons as shown in the next table.

LINESTATUS

INDICATOR

Displays the name of the account that is in use. Select another account by pressing the LINESELECTOR BUTTONS

SOFT-BUTTONS(ExcludingGXP-2000)

The soft-buttons are context sensitive and will change depending on the status of the phone.Typical functions assigned to soft-buttons are:

NEW CALL Press this button to make a new hand-free call.

FORWARD ALL Unconditionally forwards the main phone line to another phone

MISSED CALLS This option shows up there were unanswered calls to this phone. TheMissedCalls option shows a list of the missed calls

CALL RETURN Calls the phone that called/tried to call your phone last.

REDIAL Redials the last number

END CALL Hangs up the call


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SPECIALSOFTBUTTONS(ExcludingGXP2000and OnlyWhenIntegrated

withGXE5024/5028)

Call Parking:

FOR GXP2020/GXP2010 ONLY. Refer to the GXE5024/5028 Online User Manual

for more information.

CallPark When a GXP2020 dials out, the Call Park soft button will displayon screen. To park the call, press the Call Park button.

PickUp When another GXP2020 goes off-hook the Call Pickup soft buttonwill display on screen. To pickup the parked call, press the CallPickup button.

Call Queue:

SignIn Press this button to sign in to the call queue. Agent will be prompted inthe LCD display to select the call queue to join. Press menu button onkeypad to select ok. Once the agent completely signs in, the agent willbe brought back to the main screen.

FOR GXP2020/2010 and 1200 only.Refer to the GXE5024/5028 Online UserManual for more information.

SignOut Press this button to sign out of the call queue. Press menu button onkeypad to select ok. This will be displayed once the agent is signed in tothe call queue.

PUBLIC MODE(Also mentioned on p.31 of this manual): This useful mode complements theCall Queue feature by allowing various user agents to log in/log off, sharing the same phone.When enabled, all other accounts on the phone will not be active. For more information, refertohttp://www.grandstream.com/support/gxe_series/gxe502x/documents/gxe502X_call_queue_

with_gxp.pdf

LogIn Press this button to log in the user agent into the call queue.

Tab Press this button to jump to toggle between UserName and Passwordentry fields.

Backspace Press this button to erase the previously typed digit, letter, or character.

LogOut Press this button to log out the user agent out of the call queue.

Table 9: LCD Icons

Icon LCD Icon Defini tions

Connectivi ty Status / SIP Proxy/Server Icon:Solid connected to SIP Server/IP address receivedBlinking physical connection failedBlank SIP Proxy/Server not registered

Phone Status Icon:OFF when the handset is on-hookON when the handset is off-hook

Speaker Phone Status Icon:FLASH when phone rings or a call is pendingOFF when the speakerphone is offON when the speakerphone is on

DND Icon:ON when the do not disturb is activated

Activate by pressing MUTE/DEL button once
http://www.grandstream.com/support/gxe_series/gxe502x/documents/gxe502X_call_queue_with_gxp.pdfhttp://www.grandstream.com/support/gxe_series/gxe502x/documents/gxe502X_call_queue_with_gxp.pdfhttp://www.grandstream.com/support/gxe_series/gxe502x/documents/gxe502X_call_queue_with_gxp.pdfhttp://www.grandstream.com/support/gxe_series/gxe502x/documents/gxe502X_call_queue_with_gxp.pdfhttp://www.grandstream.com/support/gxe_series/gxe502x/documents/gxe502X_call_queue_with_gxp.pdfhttp://www.grandstream.com/support/gxe_series/gxe502x/documents/gxe502X_call_queue_with_gxp.pdf


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Calls Forwarded Icon:INDICATES calls are forwardedFollow call forwarding procedures

Handset, Speakerphone and Ring Volume Icon:

Each icon appears next to the volume iconTo adjust volume, use the up/down button

Realtime Clock:Synchronized to Internet time serverTime zone configurable via web browser

AM/PM indicatorPM

AM


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TABLE 10: GXPKEYPAD BUTTONS

Key Button Key Button Definitions

LINE BUTTONS Line keys with LED, can be configured to different SIP profiles

TRANSFER TRANSFER key: Transfer an ACTIVE call to another number

CONF Press CONF button to connect Calling/Called party into conference

MUTEMute an active call; or Delete a key entry

Also used to REJECT incoming call.

HOLD Place ACTIVE call on hold

MSG Enter to retrieve voice mails or other messages

Enable/Disable hands-free speaker mode

SENDPress SEND to dial a new number or redial the last number dialed. Presssend button to send a call immediately before no key entry timeout valueexpires

Enter to retrieve voice mails or other messages

MENUEnter Keypad Configuration MENU mode when phone is in IDLE mode.Use as ENTER key when in Keypad Configuration.

0 - 9, *, #Standard phone keypad; press # key to send call; press * key to for IVRfunctions

DND DO NOT DISTURB key; Press DND to turn Do not disturb function on or off.

HEADSETPress HEADSET key to answer/hang up phone calls while using headset. Italso allows user to toggle between headset and speaker. Not available onGXP2000.

INTERCOM Turn intercom function on/off

Brings phonebook on screen


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MAKING PHONE CALLS

Handset, Speakerphone and Headset ModeThe GXP series phones allow you make phone calls via handset, speakerphone, or headset mode. During

the active calls the user can switch between the handset and the speaker by pressing the speaker key. Forheadsets to operate, the user must plug the headset to an RJ22 or 2.5mm port on the phone, which allowsthe user to pick-up, speak, or hang-up calls.

Multiple SIP Accounts and LinesGXP can support up to six independent SIP accounts depending on the product model. Each account iscapable of independent SIP server, user and NAT settings. Each of theline buttons is virtually mapped toan individual SIP account. The name of each account is conveniently printed next to its correspondingbutton. In off-hook state, select an idle line and the name of the account (as configured in the web interface)is displayed on the LCD and a dial tone is heard.

For example: Configure ACCOUNT 1 and ACCOUNT 2 with Account Name as VoIP 1, VoIP 2,

respectively and ensure that they are active and registered. When LINE1 is pressed, you will hear a dialtone and see VoIP 1 on the LCD display; when LINE2 is pressed, you will hear a dial tone and see VoIP 2on the LCD display.

To make a call, select the line you wish to use. The corresponding LINE LED will light up in green. User canswitch lines before dialing any number by pressing the same LINE button one or more times. If you continueto press a LINE button, the selected account will circulate among the registered accounts.

For example: when LINE1 is pressed, the LCD displays VoIP 1; If LINE1 is pressed twice, the LCDdisplays VoIP 2 and the subsequent call will be made through SIP account 2.

Incoming calls to a specific account will attempt to use its corresponding LINE if it is not in use. When thevirtually mapped line is in use, the GXP will flash the next available LINE (from left to right or from top to

bottom for Multi Purpose Keys) in red. A line is ACTIVE when it is in use and the corresponding LED is red.

Completing CallsThere are six ways to complete a call:

1. DIAL: To make a phone call. Take Handset/SPEAKER/Headset off-hook

or press an available LINE key (activates speakerphone)or press the NEW CALL soft-key.

The line will have a dial tone and the primary line (LINE1) LED is red.If you wish, select another LINE key (alternative SIP account).

Enter the phone number Press the SEND key

or press the DIAL soft-key.

2. REDIAL:To redial the last dialed phone number.When redialing, the phone will use the same SIP account as was used for the last call. Thus, whenthe third SIP account was used for the last call/call attempt, the phone will use the third account toredial. Take Handset/SPEAKER/Headset off-hook or

press an available LINE key (activates speakerphone), the corresponding LED will be red. Press the SEND button

or press the REDIAL soft-key.


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3. CALL RETURN:To call the last phone number that called your phone.When returning a call, the phone will use the same SIP account as the call was made to. Thus, whenreturning a call made to the third SIP account, the phone will use the third SIP account return thecall.

i. Hand-free option

1. Press the CALL RETURN soft-keyii. Hand-set option1. Take the Handset off-hook2. Press the CALL RETURN soft-key

4. USING THE CALL HISTORY: To call a phone number in the phones historyWhen using the call history, the phone will use the same SIP account as was used for the lastcall/call attempt. Thus, when returning a call made to the third SIP account, the phone will use thethird SIP account return the call.

Press the MENU button to bring up the Main Menu.

Select Call History and then Received Calls, Missed Calls or Dialed Calls depending onyour needs

Select phone number using the arrow keys

Press OK to select

Press OK again to dial.

5. USING THE PHONEBOOK: Calling a phone in from the phones phonebook.Each entry in the phonebook can be attached to an individual SIP account. The phone will use thatSIP account to make the phone call.

Go to the phonebook by:

i. Pressing the phonebook button (bottom, left-hand side of phone), or

ii. Pressing the DOWN arrow key, or

iii. Pressing the menu button andSelecting Phone book andPress MENU

Select the phone number by using the arrow keys Press OK so select

Press OK again to dial.

6. PAGING/INTERCOM:The paging/intercom function can only be used if the SERVER/PBX supports this feature and boththe phones and PBX are correctly configured.

Take the Handset/SPEAKER/Headset off-hook,

Select the LINE key associated with account

Press OK key to display LCD: LINEx: PAGE USING.

Dial the phone number you want to Page/Intercom

Press SEND key.

NOTE: Dial-tone and dialed number display occurs after the handset is off-hook and the line key is selected.The phone waits 4 seconds (by default; No key Entry Timeout) before sending and initiating the call. Pressthe SEND or # button to override the 4 second delay.

Speed DialThe Multi Purpose Key buttons, located on the right-hand-side of the phone, can be configured for speeddial. Press the speed dial button to automatically call the assigned extension.


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Note: The multi-functional buttons will function as LINE keys when all LINEs are busy. The LED will flash inred to indicate an incoming call. Press the button to pick up the call. If any one of the Multi Purpose Keys isassociated with a call, the buttons speed dial/BLF function will not work.

Making Calls using IP AddressesDirect IP calling allows two phones to talk to each other in an ad hoc fashion without a SIP proxy. VoIP callscan be made between two phones if:

Both phones have public IP addresses, or

Both phones are on a same LAN/VPN using private or public IP addresses, or

Both phones can be connected through a router using public or private IP addresses (with necessaryport forwarding or DMZ)

To make a direct IP call, please follow these steps:

1. Press MENU button to bring up MAIN MENU.

2. Select Direct IP Call using the arrow-keys.

3. Press OK to select.

4. Input the 12-digit target IP address. (Please see example below).

5. Press OK key to initiate call.

To make a quick IP call, please see next section.

For example: If the target IP address is 192.168.1.60 and the port is 5062 (e.g. 192.168.1.60:5062), inputthe following: 192*168*1*60#5062 - The * key represent the dot. ; The # key represent colon :.PressOK to dial out.

Quick IP Call Mode

The GXP also supports Quick IP call mode. This enables the phone to make direct IP-calls, using only thelast few digits (last octet) of the target phones IP-number.This is possible only if both phones are in under the same LAN/VPN. This simulates a PBX function usingthe CMSA/CD without a SIP server. Controlled static IP usage is recommended.

Setting up the phone to make Quick IP calls

To enable Quick IP calls, the phone has to be setup first. This is done through the web-setup function. In theAdvanced Settings page, set the "Use Quick IP-call mode to YES. When #xxxis dialed, where x is 0-9 andxxx


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ANSWERING PHONE CALLS

Receiving Calls1. Incoming single call: Phone rings with selected ring-tone. The corresponding account LINE

flashes red. Answer call by taking Handset/SPEAKER/Headset off hook or pressing SPEAKER orby pressing the corresponding account LINE button.

2. Incoming multiple calls: When another call comes in while having an active call, the phone willproduce a Call Waiting tone (stutter tone). Next available lines will flash red (as described in section4.3.2). Answer the incoming call by pressing its corresponding LINE button. The current active callwill be put on hold.

3. Paging/Intercom Enabled: Phone beeps once and automatically establishes the call viaSPEAKER. (PBX (or Server) must also supports this feature)

Do Not Disturb1. Press the DND or MUTE button if you do not want to take a call. This will send the caller directly

to voicemail.

2. Press the DND or MUTE button to set phone to do not disturb (icon will be on the screen). Thephone will not ring and send caller directly to voicemail. (see note above)

PHONE FUNCTIONS DURING A PHONE CALL

Call Waiting/ Call Hold1. Hold : Place a call on hold by pressing the HOLD button.

2. Resume: Resume call by pressing the corresponding blinking LINE.

3. Multiple Calls: Automatically place ACTIVE call on HOLD by selecting another available LINE to

place or receive another call. Call Waiting tone (stutter tone) audible when line is in use.

Mute/Delete1. Press the MUTE button to enable/disable muting the microphone.

2. The Line Status Indicator will show LINEx: SPEAKING or LINEx: MUTE to indicate whether themicrophone is muted.

NOTE: Pressing MUTE button for an incoming call will reject the call. MUTE button also functions as deletekey when user wishes to delete the last entered digit.

Call TransferGXP supports Blind, Attended (or supervised), Semi-Attended and Auto-attended (for GXP1200 only)transfer:

NOTE: To transfer calls across SIP domains, SIP service providers must support transfer across SIPdomains. Blind transfer will usually use the primary account SIP profile.


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1. Blind Transfer: Press TRANSFER (or TRNF for GXP-2000) button, then dial the number andpress the SEND button to complete transfer of active call.

2. At tended (or Supervised) Transfer: Press LINEx button to make a call and automatically placethe ACTIVE LINE on HOLD. Once the second call is established, press TRANSFER (or TRNF)key then the LINE button of the waiting line to transfer the call. Hang up the phone call afterTransfer Successful is displayed in the screen.

3. Semi-Attended Transfer: In the web UI, under Account Settings, select the Semi-attendedTransfer Mode. There are two modes to select for this feature. RFC5589 and Send Refer with earlydialog:

a. With RFC5589 phone will not send out REFER to transferee until 200OK is received fromtransfer target. Like an attended transfer, press LINEx button to make a call andautomatically place the ACTIVE LINE on HOLD. Dial the second call, and as it is ringingpress the TRANSFER (or TRNF) key and then the blinking LINE button of the call on holdto transfer the call.

b. With "Send REFER with early dialog" mode phone will send out REFER as soon as thetransfer target is selected (in the early dialog stage). Like an attended transfer, press LINEx

button to make a call and automatically place the ACTIVE LINE on HOLD. Dial the secondcall, and as it is ringing press the TRANSFER (or TRNF) key and then the blinking LINEbutton of the call on hold to transfer the call. Transferor will hang up.

4. Auto-Attended Transfer (only on GXP1200): In the web UI, under Advanced Settings, set Auto-attended Transfer to Yes.

Establish a call between two phones

Press TRANSFER button on GXP1200

Select another line on GXP1200, call another phone and press SEND button

If the remote phone answers, press TRANSFER button on GXP1200 to transfer the call

NOTE: If there is no extra line available, GXP1200 will do the transfer as if Auto-attendedtransfer is disabled.

5-Way ConferencingGXP can host conference calls and supports up to 5-way conference calling.

2. Initiate a Conference Call :

Establish a connection with two or more parties

Press CONF button

Choose the desired line to join the conference by pressing the corresponding LINE button.

Repeat previous two steps for all other parties that would like to join the conference. Thiscan be done at any time. However, if a new call comes in, the other calls will be placed onhold and the host will have to individually re-join the held lines back into the conference by

repeating the previous two steps again.3. Cancel Conference: Canceling establishing conference call.

If after pressing the CONF button, a user decides not to conference anyone, press CONFagain or the original LINE button.

This will resume two-way conversation.

4. End Conference: Press HOLD to end the conference call and put all parties on hold; To speak with an individual party, select the corresponding blinking LINE.


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NOTE: The party that starts the conference call has to remain in the conference for its entire duration, youcan put the party on mute but it must remain in the conversation.

Voice Messages (Message Waiting Indicator)A blinking red MWI (Message Waiting Indicator) indicates a message is waiting. Press the MSG button toretrieve the message. An IVR will prompt the user through the process of message retrieval. Press aspecific LINE to retrieve messages for a specific line account.

NOTE:

Each line has a separate voicemail account. Each account requires a voicemail portal number to beconfigured in the voicemail user id field.

To check which line account has a message 1) press the message button (this always checks theprimary account), 2) check each line for stutter tone or 3) check missed calls using the menu.

Busy Lamp Field

The Multi Purpose Key buttons can be configured for Busy Lamp Field function with a specified account.When BLF is configured on one of the multi-functional buttons, the Speed Dial function will work when thatline is not in use. Call Pick Up is supported when user presses a flashing BLF key.


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CALL FEATURES

The GXP supports traditional and advanced telephony features including caller ID, caller ID w/name, callforward/transfer/park/hold as well as intercom/paging and BLF.

Table 11: GXP Call Features

Key Call Features

*30 Block Caller ID (for all subsequent calls)

*31 Send Caller ID (for all subsequent calls)

*67 Block Caller ID(per call)

*82 Send Caller ID (per call)

*50 Disable Call Waiting (for all subsequent calls)

*51 Enable Call Waiting (for all subsequent calls)

*70 Disable Call Waiting (per Call)

*71 Enable Call Waiting (per Call)

*72 Uncondi tional Call Forward

Dial *72 for a dial tone. Dial the forwarding number followed by #. Wait for dialtone. LCD will display Call FWD Activated.

*73 Cancel Uncondi tional Call Forward: dial *73 and get the dial tone, then hang up. LCD will display Call FWD Activated.

*90 Busy Call Forward

Dial *90 for a dial tone. Dial the forwarding number followed by #. Wait for a dialtone. Hang up.

*91 Cancel Busy Call Forward : dial *91. Wait for dial tone. Hang up.

*92 Delayed Call Forward

Dial *92 for a dial tone. Dial the forwarding number followed by #. Wait for a dialtone. Hang up. LCD will display Call FWD Activated.

*93 Cancel Delayed Call Forward

Dial *93 for a dial tone, then hang up.

CUSTOMIZED LCDSCREEN &XML

Grandstream GXP Series phones support both simple and advanced XML applications: 1) XML Custom Screen,2) XML Downloadable Phonebook and 3) Advanced XML Survey Application. For more information on how tocreate a downloadable XML phonebook, creating a custom idle screen and/or reprogramming the soft-keys onGXP-1200/GXP-2010/GXP2020, please visit our websiteat:http://www.grandstream.com/support/gxp_series/general/gxp_support.html.
http://www.grandstream.com/support/gxp_series/general/gxp_support.htmlhttp://www.grandstream.com/support/gxp_series/general/gxp_support.htmlhttp://www.grandstream.com/support/gxp_series/general/gxp_support.htmlhttp://www.grandstream.com/support/gxp_series/general/gxp_support.html


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Configuration Guide

The GXP can be configured in two ways. Firstly, using the Key Pad Configuration Menu on the phone; secondly,through embedded web-configuration menu.

CONFIGURATION VIA KEYPAD

To enter the MENU, press the round button. Navigate the menu by using the arrow keys: up/down and left/right.Press the OK button to confirm a menu selection, delete an entry by pressing the MUTE/DEL button. The phoneautomatically exits MENU mode with an incoming call, the phone is off-hook or the MENU mode if left idle for 20seconds.

Press the MENU button to enter the key the Key Pad Menu. The menu options available are listed in table 8.

Table 12: Key Pad Conf iguration Menu

Call History Displays histories of incoming, dialed and missed calls.

Status Displays the network status, account statuses, software version andMAC-address of the phone.

Phone Book Displays the phonebook

LDAP Directory Displays the LDAP directory

Instant Messages Goes to voice messages

Direct IP call Displays the IP-call options menu

Preference Press Menubutton to enter this sub menu including

Do NOT DisturbDND (Do NOT Disturb) function could be turned on or off inthe DO NOT Disturb menu.

Ring ToneChoose different ring tones in the Ring Tone menu.

Ring VolumePress Menubutton to hear the selected ring volume, press orto hear and adjust the ring tone volume.

LCD Contrast LCD Brightness Download SCR XML

The phone will download the custom idle screen (if available) Erase Custom SCR

Custom idle screen will be erased and will be replaced withdefault Grandstream logo.

Display LanguageYou can choose English, Chinese or Secondary Language

Press Menubutton to choose the menu item.Press to return to the main menu.

Configure Press Menubutton to display the configuration selections:

Network.To enable/disable DHCP.To setup IP-address, Net mask and Gateway address


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SIPTo change SIP-server settings for primary account.

UpgradeIn this menu setting regarding the firmware server and Configserver can be changed. It also enables the user to make the

phone attempt to download new firmware. Multi Purpose Key (On GXP2000/2010/2020 only)

To configure multi-purpose keys. Factory Reset

Key in the physical/MAC address on back of the phone.Press Menubutton to reset FACTORY DEFAULT setting. Do notuse Factory Reset unless you want to restore factory settings

Layer 2 QoSConfigure Vlan Tags

Press to return the main menu.

Factory Functions Press Menuto display the factory function items including

Audio LoopbackSpeak into the handset. If you hear your voice in the handset,your audio works fine.Press Menubutton to exit the mode.

Diagnostic ModeAll LEDs will light upPress any key on the keypad, to display the button name in theLCD. Lift and put back the handset or press Menubutton to exitthe diagnostic mode.

Enable WDTToggles the status of the Watchdog Timer.

Press to return to the main menu.

Reboot Press Menubutton to reboot the deviceDisplay ExitPress Menubutton to exit the menu

Exit Exit from this menu.


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FIGURE 3:KEYPAD GUIFLOW

Call History

Status

Phone Book

LDAP Directory

InstantMessage

Direct IP Call

Preference

Config

FactoryFunctions

Reboot

Exit

MENU

Answered Calls

Dialed CallsMissed CallsBack

Call Histor y

Back

Clear All

Any of previous menus

New EntryDownload Phonebook XMLBack

Phone BookName:Number:

Acc t:Confirm Add:Cancel & Return:

New Entry

View DirectoryDownload DirectorySearch ConfigurationBack

LDAP Directory

Select Fil terFilter ValueBack

Search Configuration

Clear All

Back

Preference

Do Not DisturbRing ToneRing VolumeLCD ContrastLCD BrightnessDownload SCR XMLErase Custom SCRDisplay Language

Back

NetworkSIPUpgradeMulti-Purpose KeysFactory ResetLayer 2 QoS

Aud io LoopbackDiagnostic Mode

Enable WDTBack

Factory Function

Config

Enable DNDDisable DND

Back

Default RingRing1Ring2Ring 3Back

Act iveIdleBack

EnglishChineseSecondary LanguageLanguage File Postfix

Back

Do Not Disturb

Ring Tone

LCD Brightness

Display Language

IP SettingIPNet MaskGatewayDNSServer1DNSServer2

Back

AccountSIP ProxyOutboundProxySIP User IDSIP Auth IDSIP PasswoSIP Transpo

Aud ioSave

Network

FirmwareServerConfig ServeUpgrade Via

802.1Q/VLAN TPriority valueReset Vlan ConBack

Upgrade

Layer 2 QoS

Instant Message

Keypad/LED Diagnosti c

Diagnostic Mode

SIP


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CONFIGURATION VIAWEB BROWSER

The GXP embedded Web server responds to HTTP/HTTPS GET/POST requests. Embedded HTML pagesallow a user to configure the IP phone through a Web browser such as Microsofts IE or Mozilla Firefox.

Access the Web Configurat ion Menu

To access the phones Web Configuration Menu

Connect the computer to the same network as the phone1

Make sure the phone is turned on and shows its IP-address

Start a Web-browser on your computer

Enter the phones IP-address in the address bar of the browser2

Enter the administrators password to access the Web Configuration Menu3

1 The Web-enabled computer has to be connected to the same sub-network as the phone. This can easilybe done by connecting the computer to the same hub or switch as the phone is connected to. In absence

of a hub/switch (or free ports on the hub/switch), please connect the computer directly to the phone usingthe PC-port on the phone.

2 If the phone is properly connected to a working Internet connection, the phone will display its IP address.This address has the format: xxx.xxx.xxx.xxx, where xxx stands for a number from 0-255. You will needthis number to access the Web Configuration Menu. e.g. if the phone shows 192.168.0.60, please usehttp://192.168.0.60 in the address bar your browser.

3 The default administrator password is admin; the default end-user password is 123.

NOTE: When changing any settings, always SUBMIT them by pressing the button on the bottom of thepage. Reboot the phone to have the changes take effect. If, after having submitted some changes, moresettings have to be changed, press the menu option needed.

Definitions

This section will describe the options in the Web configuration user interface. As mentioned, a used can login as an administrator or end-user.

Functions available for the end-user are: Status:Displays the network status, account statuses, software version and MAC-address of the

phone Basic:Basic preferences such as date and time settings, multi-purpose keys and LCD settings can

be set here.

Additional functions available to administrators are: Advanced Sett ings: To set advanced network settings, codec settings and XML configuration

settings. Account X: To configure each of the SIP accounts. EXT X:To configure setting on extension module


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Table 13: Device Configuration - Status

MAC Address The device ID, in HEXADECIMAL format.

IP Address This field shows IP address of GXP

Product Model This field contains the product model information.

Part Number This field contains the product part number

Software Version Program: This is the main software (firmware) release number, always used toidentify the software (firmware) system of the phone.

Boot: Booting code version number

System Up Time This field shows system up time since the last reboot.

System Time This field shows the current time on the phone system.

Registered Indicates whether accounts are registered to the related SIP server(s). GXP cansupport four unique SIP profiles.

PPPoE Link Up Indicates whether the PPPoE connection is enabled (connected to a modem).

Table 14: Device Configuration Basic Settings

End UserPassword

This contains the password to access the Web Configuration Menu. This fieldis case sensitive with a maximum length of 25 characters.

IP Address The GXP operates in two modes:

1. DHCP mode: all the field values for the Static IP mode are not used (eventhough they are still saved in the Flash memory.) The GXP acquires its IPaddress from the first DHCP server it discovers on its LAN. The DHCPoption is reserved for NAT router mode. To use the PPPoE feature, set thePPPoE account settings. The GXP establishes a PPPoE session if any ofthe PPPoE fields is set.

2. Static IP mode: configure all of the following fields: IP address, SubnetMask, Default Router IP address, DNS Server 1 (primary), DNS Server 2(secondary). These fields are set to zero by default.


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Multi Purpose Key X These options are used to assign a function to the corresponding multi purpose key.Options available are:

1. Speed Dial.2. BLF (Busy Lamp Field). This option has to be supported on the PBX and it

indicates the status of the extension. The three possible states are idle(green), busy (red), ringing (blinking red).

3. Presence Watcher. This option has to be supported by a presence serverand it is tied to the Do not disturb status of the phone.

4. Eventlist BLF. This option is similar to the BLF option but in this case thePBX collects the information from the phones and sends it out in one singlenotify message.

5. Speed Dial Via Active Account. This option will act just like speed dial, butbased on the current active account. For instance, if the phone is offhookand account 4 is active, it will call the configured speed dial number usingaccount 4.

Each function is connected to one of the accounts and has a target user ID.

Time Zone This parameter controls the date/time display according to the specified time zone.

LCD Backlight AlwaysOn

Turn on LCD backlight at all times. Default is No. This option applies to GXP-1200/GXP-2000 only.

Time Display Format LCD time display in 12 hour or 24 hour format

Date Display Format Choose one of the following formats: Year-Month-Day Month-Day-Year Day-Month-Year

This option applies to GXP280/GXP285/GXP1200/GXP2000 only.

Display Clock instead ofDate

Choose to display clock or date on LCD. This option applies to GXP-280/GXP-285/GXP-1200/GXP-2000 only.


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Daylight Savings Time This parameter controls time displayed in daylight savings time. If set to Yes, then thedisplayed time will be 1 hour ahead of normal time.

The Optional Rule is configured to automatically adjust the Daylight Savings Time (DST)based on the rule set in this field.

Rule Syntax: start-time; end-time; saving

Both start-time and end-time have the same syntax: month,day,weekday,hour,minute

o month: 1,2,3,..,12 (for Jan, Feb, .., Dec)

o day: [+|-]1,2,3,..,31

o weekday: 1, 2, 3, .., 7 (for Mon, Tue, .., Sun), or 0 which means the daylightsaving rule is not based on week days but based on the day of the month.

o hour: hour (0-23), minute: minute (0-59)

If weekday is 0, it means the date to start or end daylight saving is at exactly the given date.In that case, the day value must not be negative. If weekday is not zero and day ispositive, then the daylight saving starts on the first day the iteration of the weekday (e.g.:1st Sunday, 3rd Tuesday etc). If weekday is not zero and day is negative, then thedaylight saving starts on the last day the iteration of the weekday (e.g.: last Sunday, 3rd lastTuesday etc).

The saving is in the unit of minutes. The saving time may also be preceded by a negative (-)sign if subtraction is desired instead of addition.

Thedefault valueis set for US, the Automatic Daylight Saving Time Rule shall be set to 3,2,7,2,0;11,1,7,2,0;60

US/Canada where daylight saving time is applicable:Examples

03,02,7,02,00;11,1,7,02,00;60This means the daylight saving time starts from the second Sunday of March at 2AM andends the first Sunday of November at 2AM. The saving is 60 minutes.

LCD Backlight

Brightness

Set the LCD brightness level. Range from 0 to 8 where 0 means off and 8 means

the brightest. For GXP2010 and GXP2020 only.

LCD Contrast Set LCD contrast. Range from 0 to 20. Not for GXP280/285

Disable in -call DTMFdisplay

Default is No. This field is used to hide the keypad input during a call.

Mute Speaker Ringer Default is No. When its enabled, speaker wont ring on an incoming call.

Disable Missed CallBacklight

Default is No. By default, LCD backlight will lit whenever there is a missed call.Not for GXP280/285.


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HEADSET Key Mode Select either Default mode or Toggle Speaker(default)/Headset or ToggleSpeaker/Headset(default).

In Default Mode, only the speakerphone will ring for an incoming call. User canuse the headset key to pick-up, speak, and hang up calls through headset. Theheadset icon will appear on the LCD when a call is in progress.

If Toggle Speaker/Headset(default)is checked, only the headset will ring for anincoming call.

If Toggle Speaker(default)/Headset is checked, only the speakerphone will ring foran incoming call but the user can make the phone rings using the headset bypressing the HEADSET key while phone is idle. The headset icon will appear onthe idle LCD screen.

NOTE:This is not available for GXP2000

PC Port Mode For GXP1200/GXP280 only. Default is Switch. Choose to toggle between

Switch/Hub mode.

Advanced User configuration includes not only the end user configuration, but also advanced configurationsuch as SIP configuration, Codec selection, NAT Traversal Setting and other miscellaneous configuration.

Table 15: Advanced Settings

AdminPassword

Administrator password. Only the administrator can access the AdvancedSettings and Account Settings page. Password field is purposely blank forsecurity reasons after clicking update and saved. The maximum password

length is 25 characters.

G723 rate Encoding rate for G723 codec. By default, 6.3kbps rate is set.

iLBC frame size iLBC packet frame size. Default is 20ms. For Asterisk PBX, 30ms might berequired.

iLBC payload type Payload type for iLBC. Default value is 97. The valid range is between 96 and127.

Silence Suppression This controls the silence suppression/VAD feature of the audio codec G.723 andG.729. If set to Yes, when silence is detected, a small quantity of VAD packets(instead of audio packets) will be sent during the period of no talking. If set toNo, this feature is disabled.


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Voice Frames per TX This field contains the number of voice frames to be transmitted in a singleEthernet packet (be advised the IS limit is based on the maximum size ofEthernet packet is 1500 byte (or 120kbps)).

When setting this value, be aware of the requested packet time (ptime, used inSDP message) is a result of configuring this parameter. This parameter isassociated with the first codec in the above codec Preference List or the actualused payload type negotiated between the 2 conversation parties at run time.E.g., if the first codec is configured as G.723 and the Voice Frames per TX isset to 2, then the ptime value in the SDP message of an INVITE request willbe 60ms because each G.723 voice frame contains 30ms of audio. Similarly, ifthis field is set to 2 and the first codec is G.729 or G.711 or G.726, then theptime value in the SDP message of an INVITE request will be 20ms

.

If the configured voice frames per TX exceeds the maximum allowed value, theIP phone will use and save the maximum allowed value for the correspondingfirst codec choice. The maximum value for PCM is 10 (x10ms) frames; forG.726, it is 20 (x10ms) frames; for G.723, it is 32 (x30ms) frames; for

G.729/G.728, 64 (x10ms) and 64 (x2.5ms) frames respectively.

Please be careful when editing these parameters. Adjusting these parameterswill also change the dynamic jitter buffer. The GXP has a patent dynamic jitterbuffer handling algorithm. The jitter buffer range is 20 ~ 200 ms.

Grandstream recommends using the default settings provided. Grandstreamdoes not recommend adjusting these parameters if you are an average user.Incorrect settings will affect the voice quality. Please refer to the Codec FAQathttp://www.grandstream.com/pdf/FAQ-Codec.pdf for more technical detail.

Layer 3 QoS This field defines the layer 3 QoS parameter. It is the value used for IPPrecedence or Diff-Serv or MPLS. Default value is 48.

Layer 2 QoS This contains the value used for layer 2 VLAN tag. Default setting is blank.

Data VLAN Tag Default is 0. Enabling the Data VLAN filtering will help reduce the load on thephone, but it isnt necessary in most environments. This is primarily for VLANfiltering where tagged traffic will be forwarded to the DSP.

No Key Entry Timeout Default is 4 seconds.

Use # asDial Key

This parameter allows users to configure the # key as the Send (or Dial)key. If set to Yes, the # key will immediately send the call. In this case, thiskey is essentially equivalent to the (Re)Dial key. If set to No, the # key isincluded as part of the dial string.

Local RTP port This parameter defines the starting local RTP-RTCP port pair used to listen andtransmit. It is the base RTP port for channel 0. When configured, channel 0 willuse this port _value for RTP and the port_value+1 for its RTCP; channel 1 willuse port_value+2 for RTP and port_value+3 for its RTCP. The default value is5004.

Use Random Port This parameter, when set to Yes, will force random generation of both the localSIP and RTP ports. This is usually necessary when multiple GXPs are behindthe same NAT. Default is No.
http://www.grandstream.com/pdf/FAQ-Codec.pdfhttp://www.grandstream.com/pdf/FAQ-Codec.pdfhttp://www.grandstream.com/pdf/FAQ-Codec.pdfhttp://www.grandstream.com/pdf/FAQ-Codec.pdf


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Keep-alive interval This parameter specifies how often the GXP sends a blank UDP packet to theSIP server in order to keep the hole on the NAT open. Default is 20 seconds.

Use NAT IP NAT IP address used in SIP/SDP message. Default is blank.

STUN Server IP address or Domain name of the STUN server. STUN resolution result will bedisplayed in the STATUS page of the Web UI.

Firmware Upgrade andProvisioning

Default method is HTTP. Firmware upgrade may take up to 10 minutesdepending on network environment. Do not interrupt the firmware upgradingprocess.

Via TFTP Server This is the IP address of the configured TFTP server. If selected and it is non-zero or not blank, the GXP will attempt to retrieve a new configuration file or newcode image from the specified TFTP server at boot time. It will make up to 3attempts before timeout and then it will start the boot process using the existingcode image in the Flash memory. If a TFTP server is configured and a new codeimage is retrieved, the new downloaded image will be verified and then saved

into the Flash memory.

Note: Grandstream strongly recommends that the user upgrade firmware locallyin a LAN environment if using TFTP to upgrade. Please do NOT interrupt theTFTP upgrade process (especially the power supply) as this will damage thedevice.

Via HTTP Server The HTTP server URL used for firmware upgrade and configuration via HTTP.For example: http://provisioning.mycompany.com:6688/Grandstream/1.2.5.3

Here :6688 is the specific TCP port that the HTTP server is using; omit if usingdefault port 80.

Note: If Auto Upgrade is set to No, GXP will only perform HTTP download onceat boot up.

Config Server Path IP address or domain name of firmware server.

XML Config File Password(ForGXP280/GXP/285/GXP1200Only)

The XML provisioning system allows Grandstream phones to performconfiguration updates via XML configuration files. Users can set the XML configfile password in the web UI of the phone.

Firmware FilePrefix/Postfix

Default is blank. If configured, GXP will request the firmware file with theprefix/postfix. This setting is useful for ITSPs. End user should keep it blank.

Config File

Prefix/Postfix

Default is blank. End user should keep it blank.

Al low DHCP Option 43 andOption 66 to overrideserver

Default is Yes. This allows the device to get provisioned automatically.

Authent icate Conf File Default is No. If set to Yes, configuration file would be authenticated beforeacceptance. End user should use default setting.
http://provisioning/http://provisioning/


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Automatic Upgrade This function is used by ITSP. End user should NOT touch these parameters.

Default is No. Choose Yes to enable automatic HTTP upgrade andprovisioning.In Check for upgrade every field, enter the number of minutes to check theHTTP server for firmware upgrade or configuration changes. When set to No,the phone will only perform HTTP upgrade and configuration check once at bootup.

LDAP Directory IP address or domain name of LDAP script server

Phonebook XML Enable the XML phonebook via TFTP or HTTP. Define XML server path anddownload interval. When the user downloads the XML phone the manuallyentered or edited entries will not be deleted unless this option is selected to Yes.

Idle Screen XMLDownload

Enable XML Idle Screen download via TFTP or HTTP. Select whether to UseCustom Filename or not, and define the XML server path.

XML Application Enter server path for XML application. This option applies to GXP-2020 andGXP-2010 only.

Offhook Auto Dial To configure a User ID/extension to dial automatically when the phone is takenoffhook.

DTMF Payload Type This parameter sets the payload type for DTMF using RFC2833. Default is 101.

Onhook Threshhold It determines the time handset has to be down to be recognized its onhook.Default is 800ms. For GXP280/285 only.

Syslog Server The IP address or URL of System log server. This feature is especially useful for

ITSPs.

Syslog Level Select the ATA to report the log level. Default is NONE. The level is one ofDEBUG, INFO, WARNING or ERROR. Syslog messages are sent based on thefollowing events:

product model/version on boot up (INFO level)

NAT related info (INFO level)

sent or received SIP message (DEBUG level)

SIP message summary (INFO level)

inbound and outbound calls (INFO level)

registration status change (INFO level)

negotiated codec (INFO level)

Ethernet link up (INFO level) SLIC chip exception (WARNING and ERROR levels)

memory exception (ERROR level)

The Syslog uses USER facility. In addition to standard Syslog payload, itcontains the following components: GS_LOG: [device MAC address][errorcode]error messageFor example: May 19 02:40:38 192.168.1.14 GS_LOG:

[00:0b:82:00:a1:be][000]. Ethernet link is up.


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NTP server This parameter defines the URI or IP address of the NTP (Network TimeProtocol) serve. It is used to display the current date/time.

Public Mode Default is No (standard mode). Supported only on GXP1200, GXP2010, andGXP2020. This feature is designed specifically to support the Call Queue

feature implemented on Grandstreams GXE5024 or GXE5028. If set to Yes,this feature will allow other call queue agents to login/logout using their ownlogin name and password on the same phone. When enabled, all otheraccounts will not be active.

Distinctive Ring Tone Caller ID must be configured. Select a Distinctive Ring Tone 1 through 3 for aparticular Caller ID. The GXP will ONLY use selected ring tones for particularCaller IDs. For all other calls, the GXP will use System Ring Tone. Whenselected and no Caller ID is configured, the selected ring tone will be used for allincoming calls.

System Ring Tone System ring tone. Default is North American standard.Adjust system ring tone frequencies and cadences based on local telecom

standard.

Call Progress Tones Using these settings, users can configure ring or tone frequencies based onparameters from local telecom. By default, they are set to North Americanstandard.Frequencies should be configured with known values to avoid uncomfortablehigh pitch sounds.

Syntax: f1=val,f2=val[,c=on1/off1[-on2/off2[-on3/off3]]];(Frequencies are in Hz and cadence on and off are in 10ms)ON is the period of ringing (On time in ms) while OFF is the period of silence.In order to set a continuous ring, OFF should be zero. Otherwise it will ring ONms and a pause of OFF ms and then repeat the pattern. Up to three cadencesare supported.

Intercom User ID: This field is used to configure the Intercom key in the phone. For GXP2010 andGXP2020 onlyIf the phone is working with a GS GXE502X IP-PBX it can be configured in thefollowing manner:

To page an extension : [intercom feature code]+[*]+[extension number] To page a group : [paging group feature code]+[*]+[group extension]

Disable Call Waiting Default is No. If set to Yes, the call waiting feature will be disabled.

Disable CallWaiting Tone

Default is No. If set to Yes, the call waiting tone will be disabled.

Disable Direct IP Calls Default is No. If set to Yes, direct IP calls will be disabled


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Use Quick IP Call Mode Dial an IP address under the same LAN/VPN segment by entering the last octetin the IP address.

In the Advanced Settings page there is an option Use Quick IP-call mode.Default setting is No. When set to YES, and #XXX is dialed, where X is 0-9 andXXX


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Display Language Allows user to choose preferred display language in web UI and key pad UI.

Language file postfix allows the language file to have different postfixes so thephone can request a particular file. It will append an underscore "_" plus thestring in the language file postfix.

The default language file name is "gxp.lpf". If the field Language File postfixhas "NL" string in it, then the phone will request "gxp_NL.lpf" instead of"gxp.lpf."

User can only load one secondary language.Supported Secondary language: Czech, Dutch, Estonian, French, German,Italian, Polish, Portuguese, Slovak, Slovenian and Spanish.

How to set up Secondary Language:Note: This is similar to updating firmware in your local network environment.Please refer tohttp://www.grandstream.com/faqsfirmware.html#4for details.

1. Download the language packagefromhttp://www.grandstream.com/firmware.html

2. Unzip the language package3. Open the desired language zip file4. Copy gxp.lpf to the firmware server directory using your local TFTP or HTTPserver.5. Access the advanced settings of the Web GUI, set Display Language toSecondary Language6. Update and reboot the phone

GXP has up to six line appearances, each with an independent SIP account. Each SIP account requires itsown configuration page. Their configurations are identical.

Table 16: SIP Account Settings

Account Act ive This field indicates whether the account is active. The default value for theprimary account (Account 1) is Yes. The default value for the other two accountsis No.

Account Name The name associated with each account - displayed on LCD.

SIP Server SIP Servers IP address or Domain name provided by VoIP service provider.

Outbound Proxy IP address or Domain name of Outbound Proxy, Media Gateway, or SessionBorder Controller. Used for firewall or NAT penetration in different networkenvironment. If the system detects symmetric NAT, STUN will not work. ONLYoutbound proxy can provide solution for symmetric NAT.

SIP User ID User account information provided by VoIP service provider (ITSP); either anactual phone number or formatted like one.

Authent icate ID SIP service subscribers Authenticate ID used for authentication. It can beidentical to or different from SIP User ID.
http://www.grandstream.com/faqsfirmware.html#4http://www.grandstream.com/faqsfirmware.html#4http://www.grandstream.com/faqsfirmware.html#4http://www.grandstream.com/firmware.htmlhttp://www.grandstream.com/firmware.htmlhttp://www.grandstream.com/firmware.htmlhttp://www.grandstream.com/firmware.htmlhttp://www.grandstream.com/faqsfirmware.html#4


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Authent icate Password SIP service subscribers account password for GXP to register to (SIP) servers ofITSP.

Name SIP service subscribers name that is used for Caller ID display.

Use DNS SRV: Default is No. If set to Yes, the client will use DNS SRV to look up server.

User ID is PhoneNumber

If the phone has an assigned PSTN telephone number, this field should be set toYes. Otherwise, set it to No. If Yes is set, a user=phone parameter will beattached to the From header in SIP request

SIP Regist ration This parameter controls sending REGISTER messages to the proxy server. Thedefault setting is Yes.

Un-register on Reboot Default is No. If set to Yes, the SIP users registration information will becleared from the server when the phone reboots.

SIP Instance ID Default is set No. If set to Yes it will be enabled and will add reg-ID and InstanceID on contact header in the REGISTER messages. This feature is mainlyprovided for servers that don't support SIP Instance ID feature, but will still allowphones to register.

Register Expiration This parameter allows user to specify the time frequency (in minutes) that GXPrefreshes its registration with the specified registrar. The default interval is 60minutes. The maximum interval is 65,535 minutes (about 45 days).

Local SIP Port This parameter defines the local SIP port used to listen and transmit. The defaultvalue for Account 1 is 5060. It is 5062, 5064, 5066 for Account 2, Account 3 and

Account 4 respectively.

SIP Regist ration FailureRetry Wait Time

Retry registration if the process failed. Default is 20 seconds.

SIP T1 Timeout RFC 3261 SIP T1 timer. Default is 1 second.

SIP T2 Interval RFC 3261 SIP T2 timer. Default is 0.5 seconds.

SIP Transport Choose SIP Transport between UDP and TCP. Default is UDP.

Use RFC3581Symmetric Routing

Default No. When selected the phone will follow the routing procedures specifiedin RFC3581.

NAT Traversal (STUN) This parameter activates the NAT traversal mechanism. If activated (by choosingYes) and a STUN server is also specified, the phone performs according to theSTUN client specification. Using this mode, the embedded STUN client detects ifand what type of NAT/Firewall configuration is used. If the detected NAT is a Full

Cone, Restricted Cone, or a Port-Restricted Cone, the phone will use its mappedpublic IP address and port in all of its SIP and SDP messages. If the NATTraversal field is set to Yes with no specified STUN server, the GXP willperiodically (every 20 seconds or so) send a blank UDP packet (with no payloaddata) to the SIP server to keep the hole on the NAT open.

SUBSCRIBE for MWI: Default is No. When set to Yes a SUBSCRIBE for Message Waiting Indicationwill be sent periodically.


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SUBSCRIBE forRegistration Event

Default is No. This is mainly used for IMS purposes. When enabled, the terminalsshould store the Service-Route header values after successfully registered, andthereafter add a route header with the values stored in the Service-Route wheninitiating a request excluding REGISTER.

PUBLISH for Presence Enable Presence feature.

Proxy-Require SIP Extension to notify SIP server that the unit is behind the NAT/Firewall.

Voice Mail UserID When configured, user can access messages by pressing MSG button. This IDis usually the VM portal access number.

Send DTMF This parameter specifies the mechanism to transmit DTMF digit. There are 3supported modes: in audio which means DTMF is combined in audio signal (notvery reliable with low-bit-rate codec), via RTP (RFC2833), or via SIP INFO.

Early Dial Default is No. Use only if proxy supports 484 responses.

Dial Plan Prefix Sets the prefix added to each dialed number.

BLF Call-pickup Prefix Default is **. This prefix is prepended when answering call with BLF key.

Delayed Call ForwardWait Time

Time waited before the call is forward to a number or VM.Default is 20 seconds.

Enable Call Features Default is Yes. If set to No, Call transfer, Call Forwarding & Do-Not-Disturb aresupported locally provided ITSP support those features. In addition, ForwardAllsoftkey will be hidden if call feature code is disabled for Account 1.

Call Log User can choose to disable Call Log and what kind of calls to log.

Session Expiration The SIP Session Timer extension enables SIP sessions to be periodically

refreshed via a SIP request (UPDATE, or re-INVITE. Once the session intervalexpires, if there is no refresh via a UPDATE or re-INVITE message, the session isterminated.

Session Expiration is the time (in seconds) at which the session is consideredtimed out, provided no successful session refresh transaction occurs beforehand.The default value is 180 seconds.

Min-SE Defines the minimum session expiration (in seconds). Default is 90 seconds.

Caller Request Timer If set to Yes, the phone will use session timer when it makes outbound calls ifremote party supports session timer.

Callee Request Timer If selecting Yes, the phone will use session timer when it receives inbound calls

with session timer request.

Force Timer If set to Yes, the phone will use session timer even if the remote party does notsupport this feature. If set to No, the session timer is enabled only when theremote party supports this feature. To turn off Session Timer, select No forCaller Request Timer, Callee Request Timer, and Force Timer.

UAC Specify Refresher As a Caller, select UAC to use the phone as the refresher, or UAS to use theCallee or proxy server as the refresher.


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UAS Specify Refresher As a Callee, select UAC to use caller or proxy server as the refresher, or UAS touse the phone as the refresher.

Force INVITE Session Timer can be refreshed using INVITE method or UPDATE method.Select Yes to use INVITE method to refresh the session timer.

Enable 100rel PRACK (Provisional Acknowledgment) method enables reliability to SIPprovisional responses (1xx series). This is required to support PSTN inter-networking..

Account Ring Tone There are 4uniquely defined ring tones: One (1) System Ring Tone: when selected, all calls will ring with system

ring tone. Three (3) Customer Ring Tones: when selected, incoming calls from

designated account will play selected ring tone.

Ring Timeout Defines how long ring will ring when receiving a call. Default is 60 seconds.

Send Anonymous If this parameter is set to Yes, the From header in outgoing INVITE messagewill be set to anonymous, essentially blocking the Caller ID from displaying.

Anonymous Method Whether to use in the From Header or P-Asserted-Identity header.

Anonymous CallRejection

Default is NO. If set to YES, anonymous call will be rejected

Auto Answer Default is No. If set to Yes, GXP will automatically switch on speaker to answerthe incoming call. Set to Intercom/Paging mode, it will answer the call based onthe SIP info header from the server.

Al low Auto Answer by

Call-Info

If the Call-Info header contains answer-after=0, the call be answered

automatically (so called paging mode).

Turn off speaker onremote disconnect

When BYE is received, the phone will turn off its speaker automatically.

Check SIP User ID forincoming INVITE

Check the SIP User ID in Request URI. If they dont match, the call will berejected.

Refer-To Use TargetContact

Default is NO. If set to YES, then for Attended Transfer, the Refer-To header uses thetransferred targets Contact header information.

Disable Multip le MediaAttr ibute in SDP

Default is No.

Preferred Vocoder GXP supports up to 7 different Vocoder types including G.711(a/) (also knownas PCMU/PCMA), GSM, G.723.1, G.729A/B, G.726-32, iLBC, G.722 (wide-band).

Configure Vocoders in a preference list that is included with the same preferenceorder in SDP message. Enter the first Vocoder in this list by choosing theappropriate option in Choice 1. Similarly, enter the last Vocoder in this list bychoosing the appropriate option in Choice 8.

SRTP Mode Enable SRTP mode based on selection. Default is No.


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eventlis t BLF URI If a server supports this feature, user needs to configure an "eventlist BLF" URIon the service side (i.e.:[email protected])

On the GXP, under Account page, fill in the ""eventlist BLF" field with the URIwithout the domain. (i.e.: BLF1006). Under Basic Settings, please select "eventlistBLF", choose account number, monitored number, etc.

Special Feature Default is Standard. Choose the selection to meet special requirements from SoftSwitch vendors.

SAVING THE CONFIGURATION CHANGES

After the user makes a change to the configuration, press the Update button in the Configuration Menu.The web browser will then display a message window to confirm saved changes.

Grandstream recommends reboot or power cycle the IP phone after saving changes.

REBOOTING THE PHONE REMOTELY

Press the Reboot button at the bottom of the configuration menu to reboot the phone remotely. The webbrowser will then display a message window to confirm that reboot is underway. Wait 30 seconds to log inagain.
mailto:[email protected]:[email protected]:[email protected]:[email protected]


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Software Upgrade & Customization

Software (or firmware) upgrades are completed via either TFTP or HTTP. The corresponding configurationsettings are in the ADVANCED SETTINGS configuration page.

FIRMWARE UPGRADE THROUGH TFTP/HTTP

To upgrade via TFTP or HTTP, select TFTP or HTTP upgrade method. Upgrade Server needs to be set toa valid URL of a HTTP server. Server name can be in either FQDN or IP address format. Here are examplesof some valid URLs.

firmware.mycompany.com:6688/Grandstream/1.2.5.3 72.172.83.110

There are two ways to set up the Upgrade Server to upgrade firmware: via Key Pad Menu and WebConfiguration Interface.

Key Pad MenuTo configure the Upgrade Server via Key Pad Menu options, select Config from the Main Menu, then selectUpgrade. Under this sub Menu, user can edit Upgrade Server in either an IP address format or FQDNformat. Choose Save and use TFTP or Save and use HTTP to select upgrade method. Select Rebootfrom the Main Menu to reboot the phone.

Web Configuration InterfaceTo configure the Upgrade Server via the Web configuration interface, open the web browser. Enter the GXPIP address. Enter the admin password to access the web configuration interface. In the ADVANCEDSETTINGS page, enter the Upgrade Servers IP address or FQDN in the Firmware Server Path field.Select TFTP or

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